In this paper, we describe a new way of mobility support for the Internet and Intranet called "Unified Access." Different from other similar proposed technologies like Mobile IP, our scheme is more easy to use, secure, and have the full backward compatibility with existing equipments which are already used.

Certain relationships between successive packet loss patterns and packet spacing are described. To observe a successive packet loss pattern, one possible method is to investigate test packets that are generated at certain intervals (e. g. , Poisson interval, constant regular interval). However, successive packet loss strongly depends on the interval generated. If test packets are generated with long intervals, then successive loss pattern cannot be shown. Thus, in such a method, where the packet intervals may sometimes be long or short, the successiveness of the packet loss should be considered in terms of the packet spacing. To clarify the relationship between the successive packet loss and the packet spacing, we analyze data based on observation of an actual network with the loss window size as a parameter. We find that when the packet spacing is narrower, i. e. , has a shorter interval, the probability becomes higher that the packet immediately following a single packet loss would also be lost.

IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H. 323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typical case, the bandwidth is cut by 40% for eight G. 723.1-encoded voice streams through header overhead reduction and the number of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.

This paper proposes a new model which can approximate the delay time distribution in the Internet. It is well known that the delay time in communication links follows the exponential distribution. However, the earlier models cannot explain the distribution when a communication link is heavily overloaded. This paper proposes to use the M/M/S(m) model for the Internet. We have applied our model to the measurement results. This paper deals with one-way delay because it reflects the actual characteristics of communication links. Most measurement statistics in the Internet have been based on round-trip time delay between two end nodes. These characteristics are easily measured by sending sample packets from one node to the other. The receiver side echoes back the packets. However, the results are not always useful. A long distance communication link, such as a leased line, has two different fibers or wires for each direction: an incoming link, and an outgoing link. When the link is overloaded, the traffic in each link is quite different. The measurement of one-way delay is especially important for multimedia communications, because audio and video transmissions are essentially one-way traffic.

To provide users with database-like query interfaces on HTML data, several systems have been developed to extract structures from HTML pages. Among them, tree-like structures and path expressions are the most popular modeling and navigating tools, respectively. Although path expressions are straightforward in representing top-down search patterns, they provide very limited help in representing bottom-up and in-breadth search patterns. In this paper, a lattice model is proposed to store Web data. The model provides an integrated mechanism to store text, linking information, HTML hierarchy, and sequence order of HTML data. By incorporating lattice operators with comprehension syntax, we show that the query language can represent top-down, bottom-up, and in-breadth searching patterns with uniform operators. It will be also shown that lattice comprehensions can represent all operators of path expressions, except Kleen closure.

Support of collaborative work and management of spatio-temporal data has become one of the most interesting and important database applications, which is due to the tremendous progress of database and its surrounding technologies in the last decade. In this paper, we investigate the new generation database technologies that are needed to support such advanced applications. Because of the recent progress of virtual reality technology, virtual work spaces are now available. We examine a typical CSCW (Computer Supported Cooperative Work) fsystem to identify database problems that arise from it. We introduce typical approaches to database improvement based on the high-level view and the virtual reality technique. Also, in this paper, the following are introduced and discussed: the design and implementation of three- and four-dimensional spatio-temporal database systems, VRML (Virtual Reality Modeling Language) database systems, fast access methods to spatio-temporal data, and the interval-based approach to temporal multimedia databases.

The project of interconnecting CATV in Hyogo Prefecture, Japan has started since March, 1998. In this project, there are three CATV companies in Hanshin area; Kobe, Nishinomiya and Amagasaki. An ATM switch is equipped in each company and these CATVs are connected serially in the above order. Each company provides the video service to the rest of companies using the MPEG2 over ATM. Each MPEG2 stream is sent to the other two CATV companies according to the function of multicast implemented in ATM switch. In the coverage of each CATV, subscribers utilize Internet connection using cable modems as well as standard CATV broadcasting service. In this paper, we present the outline of the research project in Hyogo Prefecture, Japan, and examine the jitter processes of MPEG streams of the testbed network by the simulation. In our testbed network, cells with two types of requirement for QoS are multiplexed; cells for MPEG2 which require the real-time transmission and those for Internet packets which are much more sensitive for the cell loss ratio. We investigate the jitter processes under some scenarios and show how the jitter process is affected by the Internet traffic and the other cell streams of MPEG2. Furthermore, we study the effect of the number of ATM switches on the jitter process when more CATV networks are added serially.

An extended desktop multimedia conference system named FREDERIC (File Retrieval Engineering on Distributed EnviRonment and Interactive Communication system) has been developed for international cooperative work by sharing CAD and image data among multi-point users. This paper describes the basic network design concept of utilizing the Internet as a best-effort service and ISDN as a high-speed guaranteed service. Service system requirements and designs were developed to access common databases and collaborative work of multimedia information those are shared by customers with desktop computers and to allow remote offices to use a plant walkthrough system. The performance of the prototype system especially focused on the file transmission time which is the key factor in developing and constructing the system. By applying the image compression technology of multi-tone entropy coding, it is shown that the short time CAD data transfer to meet the requirements can be achieved.

There is increasing demand for corporate information systems that have a simple human interface and are easy to access via WWW browsers. This paper proposes WebBASE, which integrates the WWW and relational databases. Experimental evaluation shows that WebBASE offers superior performance compared to existing products. Field studies of actual WebBASE applications show that it can improve the productivity of software developers for intranet application development.

QoS of a multimedia interactive satellite communications system, which uses a satellite circuit for downlinks and a terrestrial network for uplinks, has been measured and evaluated under inferior satellite circuit conditions. Our attention in this paper is focused on TCP throughput as the most significant QoS, and experiments and analysis on TCP throughput were performed. The measured results are in good agreement with theoretical values calculated by taking congestion avoidance algorithms of TCP into consideration. As a result, we were able to verify that the TCP throughput declines radically when the low signal quality of a satellite circuit causes packet retransmission to occur. In this case, congestion avoidance works to reduce network congestion. The congestion avoidance increases the value of a retransmission timer and decreases the window size even if the network is not congested. Assuming that throughput deterioration is caused by rain fade, the total time when the throughput declines is less than 4 hours a year in Tokyo, and the results show the system can be put to practical use.

An architecture is presented for efficient and reliable delivery of multimedia contents from a primary center (PC) to secondary centers (SCs). Requested contents are delivered from the PC to the SCs through a satellite broadcast channel, or from one SC to another SC through a terrestrial channel. Cycling methods are presented that enable sharing of the contents directory of each SC. Several fundamental models and algorithms are introduced for possible consideration during the planning and design of a contents-delivery system. Simulation has shown that using both satellite broadcast and terrestrial channels for contents delivery is superior in terms of cost to the conventional use of only a satellite network.

Low bit-rate speech and audio codings are key technologies for multimedia communications. A number of coding scheme have been developed for various applications. In Internet application, good speech and audio quality at very low bit-rate (8-16 kb/s) is valuable. Two recently proposed speech and audio-coding schemes, CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction, standardized by the ITU-T in Recommendation G. 729) and TwinVQ (Transform-domain Weighted INterleave Vector Quantization, one of the candidates for MPEG-4 audio) were compared from the viewpoints of coding schemes and quality. Although there are significant differences in their basic structures and frame lengths, this paper describes that both use the same compression techniques, such as LPC (Linear Predictive Coding)-analysis pitch-period estimation and vector quantization. While CS-ACELP provides toll quality for speech at 8 kb/s, the quality it provides for music signals is insufficient. The TwinVQ transform coder is based on LPC and vector quantization and is also capable of operating at 8 kb/s. Evaluation of these two schemes in terms of their fundamental technologies, quality, delay, and complexity showed that the quality of TwinVQ for music signals is better than that of CS-ACELP, and that the quality of CS-ACELP is better for speech signals. Therefore, TwinVQ may be better suited for one-directional Internet applications, and CS-ACELP may be better for two-directional communication.

The Advanced Intelligent Network (Advanced IN) is now commercialized and the Internet is becoming popular all over the world. If these two networks were connected, the potential would exist for new services. This paper surveys and analyzes the possibility of improving both the Internet and Advanced IN with an Advanced IN-the Internet connection. Service customization, which allows customers to define their own service specifications, is one of the most important service applications for the Advanced IN. However, some issues must be resolved before that service can be offered. This paper proposes a solution in which Internet technologies are applied to the IN. We review the system architecture of Service Logic Program (SLP) definition and execution in NTT's IN for service customization. Version management and cost for delivery are the major issue for service customization with the SLP(C) creation tool. We suggest an Internet version of the SLP(C) creation tool to solve these problems. Results of the prototype shows that connecting the IN and the Internet for service customization will benefit both customers and telecommunication operators.

Quality of service requirements are satisfied conjointly by the service model, which determines how resources are shared and by network engineering, which determines how much capacity is provided. In this paper we consider the impact of the adopted charging scheme on the feasibility of fulfilling QoS requirements. We identify three categories of charging scheme based respectively on flat rate pricing, congestion pricing and transaction pricing.

We propose a hardware-based packet forwarder for multi-gigabit IP backbone networks. The conventional Internet deploys routers as a key block, but its software-controlled architecture makes it hard to scale up the packet forwarders, especially for table-lookup processes. We propose introducing a pure connectionless (CL) switching approach with a hardware-based forwarder to construct the core part of a scalable IP multi-gigabit backbone. Compared to a software-based forwarder, the table-lookup time is reduced to 100 ns by using content-addressable memory. This hardware-based pipeline implementation easily achieves a maximum forwarding performance of up to 9. 6-Gbps, or 23 million packets per second, for applications ranging from traditional best-effort IP applications to newly emerging time-critical ones. We also consider additional processing when transferring IP packets to enhance best-effort quality. This is done using selective packet-level discarding, including early packet discard and its enhancement, to achieve minimum bandwidth guaranteed service at the packet level. We discuss the IP backbone scalability issue from the viewpoint of new IP-forwarder technologies, paying special attention to connection-oriented (CO) vs. CL switching and hardware vs. software implementation. A pure CL switching solution consisting of a CL server (CLS) and a CL client (CLC) is proposed to balance the hardware- and software-based CL transport functions. As a first step to this solution, a compact CLS has been developed. It supports 600-Mbps throughput and up to 9. 6-Gbps forwarding power using a modular architecture. It was evaluated in an ATM field trial using an experimental network. The results show the effectiveness of our approach to providing enhanced best effort services.

A traffic engineering method has been developed to meet the requirements for efficient bandwidth dimensioning and for a practical and consolidated network design method. It characterizes the offered-traffic burstiness on a transit link by using time-series measurement of the aggregate traffic. It estimates future traffic characteristics based on the average traffic volume at that time which is easily derived from trend analysis, i. e. , an x% increase in bandwidth each year and gives the required link capacity. Simulation showed that the parameters estimated using this method fit the actual behavior of a network well. This method enables an appropriate bandwidth to be allocated to a transit link without having to estimate the specific traffic characteristics for each connection over the link. Once the burstiness parameter and its trend have been identified based on this method, it is possible to use a simple traffic measurement method to detect changes in network traffic and feed them back to the engineering procedure.

To construct the future multimedia network, ATM network technology and services should support cost-effective, high-speed interconnectivity and a variety of service-providing functions. Furthermore, as the infrastructure of future multimedia service, the ATM architecture should be adaptable to changes without needing replacement of its core functions and platform capabilities. This paper presents an overview of the current state of development, standardization and deployment of the ATM network service technologies and architecture concept. It also discusses the trend toward the integration of ATM technology and Internet technology. Also reported is the state of development and standardization for the individual ATM technologies and related issues, including access networks, bearer services, signalling, network middleware, and future ATM switching system technology.

This paper describes a mobile information access system based on a multi-agent architecture. With the rapid progress of wireless data communications, mobile Internet access will be more and more popular. In mobile environments, user location plays an important role for information filtering and flexible communication service. In this paper, we propose a mobile information service system where a user with a handy terminal accesses Internet in an open air to look up map information and related town information. Each user information is managed by an independent agent process. And the agent provides each user with a personal service collaborating with other applications. A map-based information service example based on this architecture is also described.

A parallel IP router that uses off-the-shelf wor-kstations and interconnecting switches is presented. This router, called CORErouter-I, is a medium-grained, functionally distributed parallel system consisting of four kinds of processors for routing, routing-table searching, servicing, and line interfacing. Also discussed are issues related to the implementation of CORErouter-I, especially in terms of routing protocol processing and packet-forwarding. Performance characteristics of CORErouter-I are also clarified through several experiments performed to evaluate maximum throughput, analyze packet-forwarding time, and estimate the effect of parallel processing on the route-flapping problem.

This paper proposes cell-attached frame encapsulation schemes in which encapsulation processing can be performed without cell reassembly. The proposed schemes are especially useful for a global networking service platform to integrate widely distributed user LANs into a single internetwork. The platform itself is an ATM-based frame forwarding network composed of access networks and a core network. These elemental networks are interconnected via edge nodes. In order to improve network interworking performance, these edge nodes should perform encapsulation processing without cell reassembly. Our proposal solves this problem. In the proposed schemes, when the first cell of a cell-divided access network frame arrives at an ingress edge node, a core-header-cell is generated from the IP header described in the first cell payload. This core-header-cell is first transmitted and then succeeding incoming cells including the first cell are forwarded cell-by-cell as soon as they arrive. Since cell-by-cell forwarding-processing reduces frame forwarding latency and cell buffer capacity, these schemes are effective from the viewpoint of both performance improvement and cost reduction.